August 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

How to set up a connection between two users by using fs_cli?
by Jacek 05 Aug '13

05 Aug '13

Hi How should it look like fs_cli command that automatically connects the call using the following method: A call to the user, when he receives a call (pick up the phone), hears the message about the implementation of the merger, and then will be automatically connected with the number of user B. I mean the shape of the script command. Is it possible to do this with *fs_cli** -**x*, or do I have to combine the script, which will implement a similar action. Cheers Sorry for my English, my native language is Polish.

2 1

Re: [Freeswitch-users] FreeSWITCH -- 30 Second call drop
by Paul 05 Aug '13

05 Aug '13

On Thu, 1 Aug, 2013 at 1:47 PM, E.J.C. Lindner <office(a)fordior.net> wrote: > Hi Paul, > > I saw you emails regarding 30 seconds call drop. I have exactly the > same issue when I want to connect my client over OpenVPN to my FS > server, which is also the vpn server at the same time (server is > on a public IP). > > All my calls are dropped after 32 seconds... very annoying and I can't > find the solution. So I'm constantly switching config for standard vs > vpn > for my client and FS (the standard config is just connecting to the > public ip which is unencrypted). > > Did you find a solution in the mean time? Please inform me, cause it's > really killing me. :=| > > I hope you can inform me on a short term; really appreciated! > > -- > Yours sincerely, > E.J.C. Lindner > Yes I found a solution to my problem. My case differs a little bit from yours in that my openvpn server is on the router, not on the PBX, in your case it's probably even a little simpler. So here is what I can advise you, 1. Ensure NAT is working properly over openVPN, so to test that all you need to do is have 2 phones on the same PBX able to call each other internally (extension 100 to extension 101 for instance). If that works then your VPN NAT is probably good. 2. Ensure that your PBX is reporting the correct external address to your vendor upstream (this is where it was messing me up). For example, for me my PBX was 10.0.0.40 (itnernal address) which the router was NATing to 24.38.231.4 (just a sample external address) and the same on the way back, when a request would come in for 24.38.231.4 it would translate it to 10.0.0.40. Because my PBX was not aware of 24.38.231.4 as it's external address, it was only listening on 10.0.0.40, it wasn't responding to the SIP ACK packets correctly, and the call was being perceived as finished and therefore hung up. To fix it in FreeSwitch I had to edit external.xml and insert the proper IP address for sip_ext_ip and rtp_ext_ip (this is from memory) I hope this helps you out cuz it cost me hours of frustration :) Paul

3 3

message/external-body as content type in SIMPLE message
by James Lotery 05 Aug '13

05 Aug '13

Hello, I'm using mod_sms in freeswitch for sending messages between registered users. One thing I'm trying to get working is sharing of images using SIP SIMPLE by setting the content-type to message/external-body (see below). However it looks like any parameters after message/external-body are being filtered out between hitting FS and reaching mod_sms. If I turn on sip trace in sofia here's what I see: ------------------------------------------------------------------------ recv 517 bytes from udp/[xxxxx]:56692 at 11:56:34.486792: ------------------------------------------------------------------------ MESSAGE sip:33979085@xxxxx SIP/2.0 Via: SIP/2.0/UDP xxx:56692;rport;branch=z9hG4bK1213589041 From: <sip:1009@xxxxx>;tag=133921508 To: <sip:33979085@xxxxx> Call-ID: 908346813 CSeq: 20 MESSAGE Content-Type: message/external-body; access-type=URL; URL=" https://xxxxxxx/51ff92f903de3_6b94294d169daab4785f.jpg" Max-Forwards: 70 Date: Mon Aug 5 13:56:01 2013 Content-Length: 0 In this example all I do in my chatplan is execute a lua script that in turn executes the send function of mod_sms without changing anything. Here is what is sent from FS : ------------------------------------------------------------------------ send 1281 bytes to udp/[xxxx]:1024 at 11:56:34.489567: ------------------------------------------------------------------------ MESSAGE sip:33979085@xxxxx:1024 SIP/2.0 Via: SIP/2.0/UDP xxxxx;rport;branch=z9hG4bKN90r1v1yvyt2j Route: <sip:33979085@xxxxx :1024>;app-id=622464153529;pn-type=google;pn-tok=APA91bH9NTVgzIyoSzaN7c4sf69gYMPd2acTQlLKEWt20_sX3acdti9EXBtpXfsHfi6RMwlq-zERhxwKmhu6pepI6rOCRZ6p5ZFC15Y-sgGyJ6HYRU93fi1SebSrsO8h679IOXSrxDTq;pn-msg-str=IM_MSG;pn-call-str=IC_MSG;pn-call-snd=ring.caf;pn-msg-snd=msg.caf;line=153ea365cc40ca1 Max-Forwards: 70 From: <sip:1009@vxxxxx>;tag=133921508 To: <sip:33979085@xxxxx :1024>;app-id=622464153529;pn-type=google;pn-tok=APA91bH9NTVgzIyoSzaN7c4sf69gYMPd2acTQlLKEWt20_sX3acdti9EXBtpXfsHfi6RMwlq-zERhxwKmhu6pepI6rOCRZ6p5ZFC15Y-sgGyJ6HYRU93fi1SebSrsO8h679IOXSrxDTq;pn-msg-str=IM_MSG;pn-call-str=IC_MSG;pn-call-snd=ring.caf;pn-msg-snd=msg.caf;line=153ea365cc40ca1 Call-ID: 55302023-3114-432b-8b90-30a5d7e66706 CSeq: 47515157 MESSAGE Contact: <sip:xxxxx> User-Agent: FreeSWITCH-mod_sofia/1.5.5b+git~20130729T215516Z~511efc5cf0 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Type: message/external-body Content-Length: 0 X-FS-Sending-Message: 9dd08617-6b31-47dd-8f6f-ff46abf521b6 As you can see everything after message/extern-body has been removed in the content type. What do I have to do so that this comes through to mod_sms intact? At which point passing through FS would this get filtered out? It is in the SIP trace but not there when it gets to mod_sms it seems. Any ideas ? Thanks in adavnce - J

1 0

H.264 OK SDP
by Jimmy Chang 05 Aug '13

05 Aug '13

Hi, We have a softphone client using H.264 codec. When we made a call to FS, we found that the OK response of FS for the INVITE missing something. There is a SDP media attribute like this from softphone client: / Media Attribute (a): fmtp:116 profile-level-id=42000C;packetization-mode=0/ But in the OK SDP of FS, we could't find this info. The softphone client would choose other resolution for this call. We know that the H.264 is a pass through video codec in FS. Our question is that is it possible and how can we add the attribute into the OK SDP? FreeSWITCH Version 1.3.17. Thanks. Jimmy

1 0

Issue with export_on_answer with Sangoma A108/A108
by DP Siddharth 05 Aug '13

05 Aug '13

Hi All, I am trying to bridge a call & want to play a sound file when legb answers. here is dialplan: <extension name="incoming"> <condition field="destination_number" expression="^4948667$"> <action application="answer"/> <action application="bridge" data="{execute_on_answer=playback /srv/sounds/oriental-insurance-deepawali-demo.wav}freetdm/1/a/9873244228"/> <action application="hangup" /> </condition> </extension> Here I am using Sangoma card & libsng_isdn library (I have tested with latest stable FreeSWITCH, wanpipe driver & libsng_isdn). FreeSWITCH: FreeSWITCH Version 1.0.head (git-9bf2726 2012-03-14 23-26-09 -0500) I have updated with v1.2.stable but same observation WANPIPE Release: 3.5.28 libsng_isdn.so.9.9.9, I have now updated it with libsng_isdn.so.7.27.2 I am getting following error: 2013-07-22 12:44:53.725215 [WARNING] ftmod_wanpipe.c:1033 [s1c2][1:2] Failed to read 4096 bytes from sangoma device: No buffer space available (-65) 2013-07-22 12:44:53.725215 [WARNING] ftdm_io.c:4059 [s1c2][1:2] raw I/O read filed 2013-07-22 12:44:53.725215 [WARNING] mod_freetdm.c:776 Failed to read from channel FreeTDM/1:2/09873244228 device 1:2! 2013-07-22 12:44:53.805207 [WARNING] ftmod_wanpipe.c:1033 [s1c2][1:2] Failed to read 4096 bytes from sangoma device: No buffer space available (-65) 2013-07-22 12:44:53.805207 [WARNING] ftdm_io.c:4059 [s1c2][1:2] raw I/O read filed I also raised this issue with sangoma ( http://support.sangoma.com/index.php?/Default/Tickets/Ticket/View/9246) but got no conclusive response in last 6 months. Kindly help in resolving same. thanks & regards, Siddharth

2 2

Audio PBX rejects video stream, kills call
by Sam Russell 05 Aug '13

05 Aug '13

Hi guys, I've been using FreeSWITCH for a couple of weeks and built a PBX and B2BUA out of it. I've seen some interesting things that happen when PBXs get m= headers that they don't like (such as Asterisk balking when it gets multiple video streams), and I've seen one recently while calling an old school PBX. Here's my build: Me(Linphone) -> PBX (FreeSWITCH) -> B2BUA (FreeSWITCH) -> Destination The B2BUA uses proxy_media mode and doesn't have a problem, but the PBX uses passthrough codecs (so it's reading the SDP packet) and sees a problem, throwing a 500 error back to my softphone The outbound packet from the PBX to the B2BUA: send 1428 bytes to udp/[210.7.45.66]:5060 at 02:46:48.407768: ------------------------------------------------------------------------ INVITE sip:+61269337555@210.7.45.66 SIP/2.0 Via: SIP/2.0/UDP 210.7.46.240;rport;branch=z9hG4bKZrKQvvHNSy64m Max-Forwards: 69 From: "Sam Russell" <sip:+6442820079@210.7.46.240>;tag=e722ggarBe7je To: <sip:+61269337555@210.7.45.66> Call-ID: 1d80cd44-781c-1231-8db7-0050568a49fc CSeq: 47499340 INVITE Contact: <sip:gw+sbc@210.7.46.240:5060;transport=udp;gw=sbc> User-Agent: FreeSWITCH-mod_sofia/1.3.13b+git~20130223T185558Z~28680c5e58 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 453 X-FS-Support: update_display,send_info Remote-Party-ID: "Sam Russell" <sip:+6442820079@210.7.46.240 >;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1375638308 1375638309 IN IP4 210.7.46.240 s=FreeSWITCH c=IN IP4 210.7.46.240 t=0 0 m=audio 32500 RTP/AVP 98 99 100 0 8 101 13 a=rtpmap:98 SPEEX/32000 a=fmtp:98 vbr=on a=rtpmap:99 SPEEX/16000 a=fmtp:99 vbr=on a=rtpmap:100 SPEEX/8000 a=fmtp:100 vbr=on a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 m=video 29910 RTP/AVP 102 103 a=rtpmap:102 H263-1998/90000 a=fmtp:102 CIF=1;QCIF=1 a=rtpmap:103 VP8/90000 The offending packet that it returns: ------------------------------------------------------------------------ recv 1357 bytes from udp/[210.7.45.66]:5060 at 02:46:48.739816: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 210.7.46.240;rport=5060;branch=z9hG4bKZrKQvvHNSy64m From: "Sam Russell" <sip:+6442820079@210.7.46.240>;tag=e722ggarBe7je To: <sip:+61269337555@210.7.45.66>;tag=teBFaBg5DjBHm Call-ID: 1d80cd44-781c-1231-8db7-0050568a49fc CSeq: 47499340 INVITE Contact: <sip:+61269337555@210.7.45.66:5060;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.2.11+git~20130723T222215Z~55b6b8424f Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, conference, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 242 X-FS-Display-Name: Outbound Call X-FS-Display-Number: sip:+61269337555@210.7.45.66 x-inin-crn: 1101488375;loc=%3cRegionDefaultLocation%3e X-FS-Support: update_display,send_info Remote-Party-ID: "Outbound Call" <sip:+61269337555@210.7.45.66 >;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 914114385 914114386 IN IP4 210.7.45.66 s=FreeSWITCH c=IN IP4 210.7.45.66 t=0 0 m=audio 17704 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 m=video 0 RTP/AVP 96 a=rtpmap:96 /0 The PBX then balks and sends a 500 error to my softphone send 704 bytes to udp/[210.7.45.66]:5060 at 02:46:48.740796: ------------------------------------------------------------------------ BYE sip:+61269337555@210.7.45.66:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 210.7.46.240;rport;branch=z9hG4bK1a68ZjKvKgKac Max-Forwards: 70 From: "Sam Russell" <sip:+6442820079@210.7.46.240>;tag=e722ggarBe7je To: <sip:+61269337555@210.7.45.66>;tag=teBFaBg5DjBHm Call-ID: 1d80cd44-781c-1231-8db7-0050568a49fc CSeq: 47499341 BYE Contact: <sip:gw+sbc@210.7.46.240:5060;transport=udp;gw=sbc> User-Agent: FreeSWITCH-mod_sofia/1.3.13b+git~20130223T185558Z~28680c5e58 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Reason: SIP;cause=488;text="Incomplete offer/answer" Content-Length: 0 ------------------------------------------------------------------------ Then I see this in the logs straight after 2013-08-05 14:46:48.736200 [DEBUG] sofia.c:1431 nua_i_media_error: unknown event 22: 988 Incomplete offer/answer The only thing odd about the SDP packet that I get back is the attribute after the m=video line. I think FreeSWITCH sees this, doesn't ignore the a= line like it should, and then gets upset with not being able to match anything before the "/0". RFC 4566 (5.13) says that "If an attribute is received that is not understood, it MUST be ignored by the receiver." So FreeSWITCH should be ignoring any "a=" attributes after it gets a "m=" line with a 0 port, but should also ignore the attribute if it can't match (but maybe leave a warning/notice in logs)? Does this look right? Has anyone else seen this, and is there a workaround? Cheers Sam

1 0

mod_dingaling <error type="modify">
by Federico Beffa 04 Aug '13

04 Aug '13

Hi, I'm having some difficulties making freeswitch talk to gmail chat. Signalling and text chat work both ways, but have troubles with audio. I'm behind NAT and use STUN to traverse it. In a problematic case I observe the following error in debug mode in the console, where user1 initiated the conversation with user2 <iq to="user1@XXX/fsvega0EC80FEA" id="313" type="error" from="user2@YYY /VtokiPhn75120521"> <ses:session type="candidates" id="3759110558" initiator="user1@XXX/fsvega0EC80FEA" xmlns:ses="http://www.google.com/session"> <ses:candidate name="rtp" address="xxx.xxx.xxx.xxx" port="23751" username="bSYbFfMybRj8D3xH" password="ri5AOommjwum60f8" preference="1.0" protocol="udp" type="stun" network="0" generation="0"></ses:candidate> <ses:candidate name="rtcp" address="xxx.xxx.xxx.xxx" port="23752" username="tVI0bVcGkAl7kdQT" password="ay23gfVzUPdaKDW4" preference="1.0" protocol="udp" type="stun" network="0" generation="0"></ses:candidate> </ses:session> <error type="modify"> <sta:bad-request xmlns:sta="urn:ietf:params:xml:ns:xmpp-stanzas"></sta:bad-request> <sta:text xml:lang="en" xmlns:sta="urn:ietf:params:xml:ns:xmpp-stanzas"></sta:text> </error> </iq> It appears that user2 would like to modify the session, but mod_dingaling does not take any action and I do not see any RTP traffic with wireshark. Is this expected? A second question is: In the session setup log I see mod_dingaling advertising a port which is outside the range specified with the configuration variables rtp-start-port and rtp-end-port in switch.xml. Do the latter only work with SIP? Thanks for any advise, Fede

1 0

Skype "Problem witch Audio Playback" - with skypopen-oss module.
by Jacek 04 Aug '13

04 Aug '13

Hi I have a problem with skypopem-oss kernel module. After installation, the audio is normal, the next time, the system starts, Skype has a "problem with audio playback", and do not make any calls. FreeSWITCH version: User-Agent: 20130802T191223Z FreeSWITCH-mod_sofia/1.2.12 git ~ ~ 1e3bfef390 Log in failed calls via Skype: http://pastebin.com/raw.php?i=eKWFGDUy Kernel & Skypopen: modinfo skypopen filename: /lib/modules/3.9.9-gr2/misc/skypopen.ko license: Dual BSD/GPL author: Original: Alessandro Rubini, Jonathan Corbet. Modified by: Giovanni Maruzzelli for FreeSWITCH skypopen srcversion: 8AA469D7BEB12A38BBDDA0C depends: vermagic: 3.9.9-gr2 SMP preempt mod_unload modversions REFCOUNT STACKLEAK_PLUGIN GRSEC parm: skypopen_major:int parm: skypopen_minor:int parm: skypopen_nr_devs:int Is it possible to work mod_skypopen by Alsa sound, using mod_alsa or mod_portaudio? I have in the office server that is 6 lines Skype, the old Freeswitch 1.0.4 everything worked, with the new is a problem. Perhaps the module does not work very stable due to the presence of an enhanced kernel Grsecurity & Pax, maybe the problem is the Gentoo Hardened and hardened gcc-compiler with PIE & SPP - gcc version 4.6.3 (Gentoo Hardened 4.6.3 p1.5, pie-0.5.2) , any case, I would prefer the whole shall use Alsa instead skypopen module. An additional reason is that Microsoft blocked Skype 1.4, and at any time can block the use of Skype 2.0, and this is the latest version of the OSS. For ALSA, even the latest version of Skype 4.1 works correctly. 6, or even 40 Skype installation can be done on separate user accounts in separate folders to $HOME, each with a separate $HOME / .asoundrc, and a separate session dbus. That's how it worked for me on an earlier version of FreeSWITCH By the way I would advise to use the Skype communication via dbus, rather than X11, it is much more stable solution, and more predictable, taking into account the Xorg will soon be replaced by Wayland and Mir, and transportation Skype X11 may stop working. Cheers Sorry for my English, my native language is Polish.

2 2

Call Intercept
by Frank Busalacchi Jr 04 Aug '13

04 Aug '13

Hi everyone! First and foremost, thanks for a great piece of software, and all the time you guys still put into this program day in and day out. Thank you! I'm having difficulty implementing an "intercept" system, and I'm thinking that I have to be "conceptually" misunderstanding something. Towards those ends, I'm hoping someone can get me back on track, I'm sick of banging my head on the wall to try to make this work. So here is my scenario: 1. A call (Call A) comes into the server from my voip provider on a DID. 2. My XML dialplan transfers the call to an extension in the XML dialplan of 1111. 3. My XML dialplan recognizes 1111 as a local extension, and executes a LUA script to handle the actual bridging of Call A to the appropriate devices. I use the LUA script because I do some logic to decide whether to ring just the user's SIP phone, the user's cell phone, or a combination of both simultaneously. Bottom line is that the LUA script does something like this after building the originate string in dialString: session2 = freeswitch.Session(dialString); if session2:answered() == true then freeswitch.bridge(session,session2); return true; else return false; end ---All of the above works as expected --- My issue is that when the LUA script starts ringing the sip phone registered as 1111, I want to 'intercept' that call from a different extension. So here is what I do: 1. Extension 1112 hears that his co-worker's extension 1111 is ringing, and wants to answer it. He dials **1111 2. My XML dialplan recognizes the request to pickup/intercept the call ringing at 1111, and calls a LUA script to figure out the uuid of the ringing phone. 3. The LUA script sets a channel variable with the UUID it choose as the appropriate candidate to intercept, and exits. 4. The XML dialplan resumes control, and does a <action application="intercept" data="${intercept-uuid}"/> . 5. The calls aren't intercepted/bridged. A pastebin of the log is here: http://pastebin.freeswitch.org/21267 What am I missing conceptually here? Thanks, Frank Frank Busalacchi Jr President Direct (714) 254-2612 [Description: C:\Users\frankjr\Downloads\McPeekEmailLogo.jpg]

4 4

Freeswitch installation
by Ashish Mishra 04 Aug '13

04 Aug '13

Sir i have downloaded the msi file for windows of freeswitch and also ran the setup file to install freeswitch...but when i double click on the freeswitchconsole.exe file from the windows explorer it automatically closes...kindly help with this... Ashish mishra

3 7